System and method for dynamically establishing optimum audio quality in an audio conference

ABSTRACT

A system and method for dynamically establishing optimum audio quality in an audio conference is disclosed. A connection with one or more remote communication devices is initially established. An available data rate associated with the connection is then determined. Next, a bandwidth is assigned based on the available data rate. Finally, the assigned bandwidth is adjusted according to the available data rate.

CROSS-REFERENCES TO RELATED APPLICATIONS

The present application claims priority from Provisional PatentApplication Ser. No. 60/360,984, filed Mar. 1, 2002, which isincorporated herein by reference in its entirety. The presentapplication is also a continuation in part of patent application Ser.No. 10/335,108, entitled Method and Apparatus for Wideband Conferencing,filed Dec. 31, 2002.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention relates generally to the field ofteleconferencing, and more particularly to a system and method fordynamically establishing optimum audio quality in an audio conference.

2. Background of the Invention

The telecommunications industry is constantly creating alternatives totravel for reaching a meeting forum. Teleconferencing has enabled manyusers to avoid long and expensive trips merely to meet with others todiscuss business related topics and make important decisions. Inaddition, teleconferencing often replaces face to face meetingsinvolving even the shortest of trips, such as those involving officelocations relatively close in distance.

Typically, teleconferencing efficiency increases as the quality of theaudio increases. Unfortunately, the quality of the audio inteleconferencing can be compromised by quality of conventional telephonelines. Telephone lines often vary markedly from one telephone line toanother telephone line. Consequently, the data rates that can beachieved utilizing the telephone lines vary considerably as well. Thevarying telephone lines and data rates that can be achieved isparticularly of concern with respect to international and/orlong-distance connections, since the variation of the telephone linesand the data rates creates a potentially unreliable communicationsystem.

Further, obtaining the best audio quality for a particular connection iscomplicated by the fact that the connection type (e.g., long-distance,international, etc.) is not typically known until the actual connectionis established between two communication devices. In addition, whenspeech compressors are utilized in an attempt to improve audio quality,matching the data rate of the speech compressors must be considered aswell.

Wideband audio-over-POTS (plain old telephone system) combines a modemwith a codec in order to send compressed speech over a phone line. Thesewideband audio-over-POTS systems are often used in broadcasting to sendhigher-quality audio over convention telephone lines. Widebandaudio-over-data (such as IP or ISDN) systems are also available andoperate similarly. However, these systems are still limited by thecommunications line, itself, and thus cannot send audio at a faster datarate or at a higher bandwidth than the communication lines canaccommodate.

Codecs (coder/decoder) compress speech into data for transmission,sometimes via conventional telephone lines. While the compression of thespeech allows for a higher quality transmission of audio data, thebandwidth of the audio data is fixed by the codec. The data rate is alsodependent upon the bandwidth, and thus, in these embodiments both thedata rate and the bandwidth are static. For instance, G.711 provides a3.3 kHz bandwidth codec capable of transmitting data at 64 kbps.

Alternatively, multi-rate codecs are capable of operating at differentrates. In other words, multi-rate codecs provide a fixed audiobandwidth, but different quality levels depending on data rates. Forinstance, G.722 can provide 7 kHz audio bandwidth capable oftransmitting data at either 48 kbps or 64 kbps. As another example,G.722.1 provides 7 kHz audio bandwidth and can transmit data from 24kbps to 32 kbps. Although varying data rates are provided for eachbandwidth, the audio bandwidth is static. Accordingly, data ratesoutside of those data rates specifically prescribed by the particularaudio bandwidth cannot be achieved. Furthermore, in wideband-over-POTSsystems, codecs are typically disabled when the data rate drops below acertain level, and narrowband audio is utilized instead to provideaudio. Thus, codecs are not practical when acceptable audio cannot beprovided due to lack of availability of a specific data rate via aconventional telephone line.

Therefore, it can be appreciated that there exists a need for a systemand method for dynamically establishing optimum audio quality in anaudio conference.

SUMMARY OF THE INVENTION

The present invention provides in various embodiments a system andmethod for dynamically establishing optimum audio quality in an audioconference.

In a system according to one embodiment of the present invention, alocal communication device establishes a connection with one or moreremote communication devices for conducting the audio conference. A datarate monitor module determines an available data rate associated withthe connection. Next, a bandwidth adjustment module assigns a bandwidthbased on the available data rate, and adjusts the bandwidth according toany changes in the data rate.

In a method according to another embodiment of the present invention, aconnection is established with one or more remote communication devicesfor conducting the audio conference. An available data rate associatedwith the connection is then determined. Next, a bandwidth is assignedbased on the available data rate by a codec. Subsequently, the bandwidthis adjusted according to any changes in the available data rate. Thisadjustment may be an increase in bandwidth according to an increase inthe available data rate or a decrease in bandwidth according to adecrease in the available data rate.

A further understanding of the nature and advantages of the inventionsherein may be realized by reference to the remaining portions of thespecification and the attached drawings.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a schematic diagram illustrating a local communication deviceestablishing a connection with a remote communication device inaccordance with one embodiment of the present invention;

FIG. 2 is a schematic diagram illustrating exemplary componentsassociated with the local communication device and/or the remotecommunication device in accordance with the present invention;

FIG. 3 is a schematic diagram illustrating exemplary modules associatedwith a connection management engine in accordance with the presentinvention;

FIG. 4 is a schematic diagram illustrating exemplary relationshipsbetween bandwidth and data rate in accordance with the presentinvention; and

FIG. 5 is a flowchart illustrating an exemplary process for dynamicallyestablishing optimum audio quality in accordance with the presentinvention.

DESCRIPTION OF THE EXEMPLARY EMBODIMENTS

As shown in the exemplary drawings wherein like reference numeralsindicate like or corresponding elements among the figures, embodimentsof a system and method according to the present invention will now bedescribed in detail. The following description sets forth an example ofa system and method for dynamically establishing optimum audio qualitybetween communication devices.

Referring now to FIG. 1, a schematic diagram illustrating a localcommunication device 102 establishing an audio connection 106 with aremote communication device 104 in accordance with one embodiment of thepresent invention is shown. The local communication device 102 and theremote communication device 104 may be a telephone, a speakerphone, aconferencing system, such as audio, video, data, multimedia, and so on,a bridge, or an audio device for use with external systems, speakers,microphones, etc. There may be more than one remote communication device104 with which the local communication device 102 establishes the audioconnection 106. For instance, the local communication device 102 canestablish an audio connection 106 with more than one remotecommunication device 104 utilizing a bridge. Further, any type of audioconnection 106 is within the scope of the invention. For example, theaudio connection may be a POTS connection, an IP connection, an ISDNconnection, a DSL connection, a satellite connection, and so on.

The local communication device 102 typically establishes the audioconnection 106 with the remote communication device 104 for the purposeof conducting an audio conference. Optimum audio quality is dynamicallyestablished by the local communication device 102 and/or the remotecommunication device 104 via the audio connection 106. The localcommunication device 102 and/or the remote communication device 104monitor the audio connection 106, and can adjust bandwidth based oncharacteristics associated with the audio connection 106, such as datarate. Accordingly, the bandwidth can slowly increase or decreaseaccording to the data rate available, resulting generally in low noise,low distortion, and optimum audio quality. Furthermore, optimum audioquality is dynamically established and maintained throughout the ongoingaudio conference by virtue of the ability of the local communicationdevice 102 and/or the remote communication device 104 to adjust thebandwidth of the audio connection 106.

Referring now to FIG. 2, a schematic diagram illustrating exemplarycomponents associated with the local communication device 102 and/or theremote communication device 104 in accordance with the present inventionis shown. As discussed herein, the local communication device 102 and/orthe remote communication device 104 may include various components.However, for simplicity of discussion, the components will be discussedin connection with the local communication device 102. It should benoted that not all elements of the local communication device 102 and/orthe remote communication device 104 are necessary in alternativeembodiments or additional elements may be included in alternativeembodiments.

A data input/output component 202 can receive audio data from a sourceand also perform audio data output functions for audio data receivedfrom the remote communication device 104. For instance, the datainput/output component may include a microphone for collecting audiodata and a speaker for outputting audio data. Audio data from at leastone microphone is forwarded to a codec 204 via a data input/outputcomponent 202 for compression of the audio data. Preferably, the codec204 can be operated at different bandwidths as well as at different datarates. In other words, the codec 204 is a codec designed such that for aconstant level of quality, a required data rate will be reduced as thebandwidth is reduced. In order to create the codec 204, for example, aconstant algorithm can be employed with the coding parameters of thealgorithm adjusted to achieve the reduction in bandwidth according tothe reduction in data rate. Alternatively, various codecs 204 may beselected depending upon the data rate available, a fixed narrowbandwidth codec 204 can be combined with a variable-bandwidth codec 204,etc. However, any codec 204 is within the scope of the invention.

The compressed audio data is forwarded to a modem 206 and/or a channeladaptor 208. Subsequently, the modem 206 and/or the channel adaptor 208forward the audio signals to an interface 212, which sends the audiosignals to the one or more remote communication devices 104 via theaudio connection 106. The modem 206 converts the audio data into ananalog signal for transmission via a POTS audio connection 106, a cableaudio connection 106, etc. The modem 206 can establish the frequency atwhich the data will be transmitted via the audio connection 106.

Similarly, the channel adaptor 208 converts the digital data into aformat acceptable for transmission via a data channel (the particularaudio connection 106 (FIG. 1)), such as IP, ISDN, DSL, and so on.However, the compressed audio data forwarded to the channel adaptor 208is not converted into an analog signal. Rather the audio data forwardedto the channel adaptor 208 is transmitted in digital form via a digitaltransmission medium, such as the aforementioned digital transmissionmediums.

A connection management engine 210 is coupled to the codec 204, themodem 206, and the channel adaptor 208 for monitoring the available datarate of the audio connection 106 and instructing the codec 204 to makeany necessary adjustments to bandwidth in order to establish optimumaudio quality. The interface 212 is coupled to the connection managementengine 210, which can monitor the rate of the connection via theinterface 212. The data rate can be monitored continuously, periodically(e.g., every five minutes), at a function specific time (e.g. at thebeginning of the conference, during the first ten minutes of theconference, etc.), and so on. Accordingly, the bandwidth may beincreased or decreased smoothly, avoiding significant noise anddistortion. Furthermore, by dynamically establishing and maintainingoptimum audio quality in this manner, potential improvements in theaudio quality below an arbitrary data rate need not be relinquished dueto an inability to adjust the bandwidth.

Referring now to FIG. 3, a schematic diagram illustrating exemplarymodules associated with the connection management engine 210 inaccordance with the present invention is shown. Optionally, an audioquality feedback module 302 can receive feedback from the localcommunication device 102, the remote communication device 104, and/orusers participating in the audio conference. For instance, the audioquality feedback module 302 may ascertain audio quality from monitoringthe audio connection 106, the modem 206, and/or the channel adaptor 208,and/or the data received via the audio connection 106 by examining thebit error rate (BER), the data rate, etc.

Alternatively, or in addition to monitoring the audio connection 106,the modem 206, and/or the channel adaptor 208, an input mechanism (notshown) may be associated with the local communication device 102 and/orthe remote communication device 104 allowing audio conferenceparticipants, or users of the devices, generally, to provide feedback asto the audio quality of the audio conference. For example, a user may beable to rate the quality of the audio conference, which causes the localcommunication device 102 and/or the remote communication device 104 toinitiate modifications in order to improve the quality in response tothe user input.

A data rate module 304 monitors the audio connection 106 for theavailable data rate associated with the audio connection 106. Differentaudio connections 106 can support varying data rates. The data ratemodule 304 determines which data rate(s) is available for sending andreceiving data. The data rate module 304 can optionally provide thisdata rate(s) information to the audio quality feedback module 302.

The data rate module 304, optionally, forwards to a data transmissionstandard module 308 and/or a bandwidth adjustment module 306 any effectsthe modem 206, the channel adaptor 208, and/or the audio connection 106have on the available data rate. Alternatively, the data rate module 304can forward the available data rate directly to the bandwidth adjustmentmodule 306.

The data transmission standard module 308 can advise the bandwidthadjustment module 306 and/or the data rate module 304 of standardbandwidths and frequencies that, typically, are associated withspecified data rates. The data transmission standard module 308 can,accordingly, be a reference for the bandwidth adjustment module 306and/or data rate module 304.

Alternatively, or in addition to being a reference, the datatransmission standard module 308 can create instructions to forward tothe bandwidth adjustment module 306 based on information received fromthe data rate module 304. In other words, in one embodiment of thepresent invention, the data transmission standard module 308 creates andforwards a command to the bandwidth adjustment module 306 to adjust thebandwidth of the audio connection 106 based on the available data rate,changes in the available data rate, etc., which the bandwidth adjustmentmodule 306 in turn forwards to the codec 204 (FIG. 2) to adjust thebandwidth.

The bandwidth adjustment module 306 can adjust the bandwidth of theaudio connection 106 and/or the frequency of the audio signal beingtransmitted via the audio connection 106 by forwarding instructions tothe codec 204 to adjust the bandwidth, as discussed herein. Forinstance, as discussed previously, the codec 204 may initially assign abandwidth to the audio connection and this bandwidth may be adjusted bythe codec 204 based on instructions received from the bandwidthadjustment module 306 as the available data rate changes.

The frequency of the audio signal may also be adjusted by the modem 206and/or channel adaptor 208 as the bandwidth and/or data rate changes.The bandwidth adjustment module 306 can instruct the codec 204 to adjustthe bandwidth based on information from the audio quality feedbackmodule 302, information from the data rate module 304, and/orinformation and/or a command from the data transmission standard module308. As discussed herein, the bandwidth may be established atcommencement of the audio conference and/or at any time during the audioconference.

In one embodiment of the present invention, the connection managementengine 210 includes a modem training module 310 for adjusting thetraining time of the modem. In this embodiment, a user of the localcommunication device 102 and/or the remote communication device 104 canadjust the training time of the modem 206 associated with the particulardevice. For example, the user can select between a short training time,which yields lower data rates, or a long training time, which yieldshigher data rates. Accordingly, the audio connection 106 can beoptimized for specific data transmissions. Any adjustment to thetraining time of the modem 206 is within the scope of the presentinvention.

Referring now to FIG. 4, a schematic diagram illustrating exemplaryrelationships between bandwidth and data rate in accordance with thepresent invention is shown. The bandwidth 402 and/or frequency range isindicated along the x-axis. The y-axis shows varying data rates 404. Anyspectrum of bandwidths 402 and/or data rates 404 are within the scope ofthe present invention. The data rates 404 are, typically, a function ofthe audio connection 106 (FIG. 1), such as POTS, DSL, etc.

Generally, an audio connection 106 is established by the localcommunication device 102 with the one or more remote communicationdevices 104, such as POTS, ISDN, etc. Once the audio connection 106 typeis agreed upon, the local communication device 102 and the one or moreremote communication devices 104 negotiate to select the codec 204(e.g., G.722, G.722.2, etc.) that both the local communication device102 and the one or more remote communication devices 104 want to utilizefor communications. A set of operational parameters, such as data rate,audio bandwidth, and so on are also negotiated between the localcommunication device 102 and the one or more remote communicationdevices 104. This type of negotiation is a common part of conferencingprotocols, such as H.323, etc. Once the local communication device 102and the one or more remote communication devices 104 have agreed uponthe aforementioned factors, the audio connection 106 is established.

Line A 406 indicates that as the data rate 404 increases, the bandwidth402 increases. Similarly, as the data rate 404 decreases, the bandwidth402 decreases. Thus, when the data rate 404 increases or decreases, amessage is forwarded to the bandwidth adjustment module 306 by the datarate module 304 (FIG. 3) and/or via the data transmission standardmodule 308 to increase or decrease the bandwidth, respectively. Asdiscussed herein, the bandwidth adjustment module 306 creates andforwards a command to the codec 204 to change the bandwidth according toan exemplary embodiment.

For example, although the audio connection 106 can achieve a data rateof 16 kbps, the data rate module may change the data rate to 12 kbps inorder to lower the BER and accomplish higher data integrity. Thebandwidth 402 may also be decreased by the bandwidth adjustment module306 according to the change in the data rate 404 from 16 kbps to 12kbps. Decreasing or increasing the bandwidth in response to a decreaseor increase in the data rate appears to occur in a linear fashion. Thelinear changes to the bandwidth eliminate the “all or nothing”characteristics of many existing systems (e.g., audio-over-data,audio-over-POTS, etc.). Furthermore, the bandwidth changes smoothly sothat the changes are not easily detected. In one embodiment, when thedata rate becomes extremely low, uncoded narrowband audio can betransmitted, the change being made smoothly so as to not be easilydetected.

Line B 408 indicates that the bandwidth 402 is only mildly increased inresponse to major increases in available data rate 404. Line A 406 andLine B 408 are representative of two different audio connection 106and/or modem 206 types, which are typically known at the beginning ofthe audio conference. Line B 408 may be utilized rather than Line A 406in a scenario in which more bits are utilized for the same audioquality, for instance. For example, if the audio connection 106 isestablished via ISDN, the available data rate 404 may be higher than theavailable data rate 404 provided by other audio connections 106. Thus, asimpler codec 204, such as G.722 may be utilized that requires lesscomputation, but the G.722 codec 204 typically uses more bits in orderto accomplish the same audio quality as that provided by a G.722.2 codec204.

As another example, Line B 408 may be utilized instead of Line A 406where the audio connection 106 is a digital channel having a portionreserved for data and a portion reserved for audio and a very high dataerror rate is associated with the audio connection 106. Thus, a largeportion of the audio connection 106 is dedicated to error correction,leaving less room available for transmitting audio data due toadditional room utilized for data error control.

Line C 410 indicates that the bandwidth 402 is left unchanged when theavailable data rate 404 changes. Any change in the bandwidth 402 inresponse to changes in the available data rate 404 is within the scopeof the present invention.

Typically, at lower frequencies, increased bandwidth largely improvesintelligibility and perceived quality of the audio signal exchangedduring the audio conference. For example, 4 kHz bandwidth audio signalsare markedly clearer than 3 kHz bandwidth audio signals, while 5 kHzbandwidth audio signals are better than 4 kHz bandwidth audio signals,but by a lesser degree, and so on. Accordingly, when the codec 204adjusts the bandwidth downward (i.e., decreasing the bandwidth) fromhigher frequencies, the audio signals typically become clearer. Inaddition, the codec 204 can conceal telephone line noise, buzzing, andso forth, thereby enhancing comprehensibility of the audio signals.

Referring now to FIG. 5, a flowchart illustrating an exemplary processfor dynamically establishing optimum audio quality in accordance withthe present invention is shown. The flowchart is discussed from theperspective of the local communication device 102 (FIG. 1). At step 502,a connection with one or more remote communication devices 104 isestablished for conducting an audio conference. The connection may bevia a modem through a POTS connection, via a data channel such as IP,ISDN, satellite, or any other transmission medium. The one or moreremote communication devices may be a telephone, speakerphone,conference system (such as audio, video, data, multimedia, etc.), abridge further coupled to at least one remote communication device 104,an audio device for use with external systems, microphones, speakers,etc.

At step 504, an available data rate associated with the connection isdetermined. The available data rate is typically limited by the linecharacteristics of the audio connection 106 (FIG. 1). For instance, someaudio connections 106 can only support data rates of 16 kbps, 20 kbps,30 kbps, and so on, while poorer audio connections 106 can support 6kbps, 8 kbps, 11 kbps, etc. Further, the data rate of the audioconnection 106 can be changed according to performance of the modem 206or the channel adaptor 208 over the audio connection 106.

At step 506, a bandwidth is assigned based on the available data rate.Typically the codec 204 assigns the bandwidth based on the availabledata rate associated with the audio connection 106. Then, at step 508,the assigned bandwidth is adjusted according to changes to the availabledata rate. The bandwidth can be assigned at the beginning of the audioconference as well as adjusted at the beginning of the audio conference,during the audio conference, etc. The bandwidth may increase as the datarate increases and vice versa in one embodiment of the presentinvention. In another embodiment of the present invention, the bandwidthmay not change as the data rate increases or decreases.

The above description is illustrative and not restrictive. Manyvariations of the invention will become apparent to those of skill inthe art upon review of this disclosure. The scope of the inventionshould, therefore, be determined not with reference to the abovedescription, but instead should be determined with reference to theappended claims along with their full scope of equivalents.

1. A method for dynamically establishing optimum audio quality in anaudio conference, comprising; establishing a connection, by a localcommunication device, with one or more remote communication devices forconducting the audio conference; determining an available data rateassociated with the connection; assigninig an audio bandwidth based onthe available data rate; and adjusting the audio bandwidthproportionally to changes to the available data rate; wherein assigningthe bandwidth further comprises adjusting a training time associatedwith the audio bandwidth.
 2. The method of claim 1, wherein theadjusting occurs at the beginning of the audio conference.
 3. The methodof claim 1, wherein the adjusting occurs during the audio conference. 4.The method of claim 1, wherein the one or more remote communicationdevices is a conferencing system, a bridge or a telephone.
 5. The methodof claim 1, wherein the connection is an analog connection or a digitalconnection.
 6. The method of claim 1, wherein the audio bandwidth isadjusted in response to received audio quality feedback.
 7. The methodof claim 1, wherein the audio bandwidth is adjusted to narrowband.
 8. Adevice in a conference system for dynamically establishing optimum audioquality, wherein the conference system is coupled to a remote device viaa connection, the device comprising: a local interface having an audioinput and an audio output; a communication module for communicating viathe connection; a codec coupled to the local interface for encoding theaudio input with a first bandwidth and decoding to generate the audiooutput; and a connection management module coupled to the codec and thecommunication interface, wherein the connection management module isoperable to, determine the first audio bandwidth proportional to theavailable data rate through the connection; and adjust the audiobandwidth proportionally to changes to the available data rate throughthe connection.
 9. The device of claim 8, wherein the communicationmodule includes: a modem coupled to the codec for modulating the encodedaudio input; and a plain old telephone service (POTS) interface coupledto the modem, wherein the POTS interface is coupled to the remote devicevia the connection, wherein the connection is POTS connection.
 10. Thedevice of claim 8, wherein the communication module includes: a channeladaptor coupled to the codec; and a digital interface coupled to thechannel adaptor and further coupled to the remote device via theconnection, wherein the channel adaptor is operable to convert theencoded audio input for transmission through the digital interface andthe connection.
 11. The device of claim 10, wherein the connection is anIP connection, an ISDN connection or a DSL connection.
 12. The device ofclaim 8, wherein the connection management module includes: an audioquality feedback module coupled to the local interface; a data ratemodule coupled to the local interface; a data transmission standardmodule coupled to the local interface; an audio bandwidth adjustmentmodule coupled to the local interface; and a modem training modulecoupled to the local interface and the communication interface module.13. The device of claim 8, wherein the connection management module isoperable to adjust the audio bandwidth at the beginning of an audioconference or during an audio conference.
 14. The device of claim 8,wherein the one or more remote devices is a conferencing system, abridge or a telephone.
 15. A conference system for dynamicallyestablishing optimum audio quality, the system comprising: a localcommunication device including: a local interface having an audio inputand an audio output; a communication module for communicating via aconnection; a codec coupled to the local interface for encoding theaudio input with a first bandwidth and decoding to generate the audiooutput; and a connection management module coupled to the codec and thecommunication interface; a remote communication device including: aremote interface having an remote audio input and an remote audiooutput; a remote communication module for communication; a remote codeccoupled to the remote interface for encoding the remote audio input withthe second audio bandwidth and decoding with the first audio bandwidthto generate the remote audio output; and a remote connection managementmodule coupled to remote codec and the remote communication interface;and a connection coupling the local communication device and the remotecommunication device, wherein the connection management module in thelocal communication device is operable to, determine the fist audiobandwidth proportional to the available data rate through theconnection; and adjust the audio bandwidth proportionally to changes tothe available data rate through the connection.
 16. The system of claim15, wherein the remote communication module includes: a remote channeladaptor coupled to the remote codec; and a remote digital interfacecoupled to the remote channel adaptor and further coupled to the localdigital interface via the connection.
 17. The system of claim 15,wherein the communication module in the local communication deviceincludes: a modem coupled to the codec for modulating the encoded audioinput; and a plain old telephone service (POTS) interface coupled to themodem, wherein the POTS interface is coupled to the remote device viathe connection, wherein the connection is POTS connection.
 18. Thesystem of claim 15, wherein the communication module in the localcommunication device includes: a channel adaptor coupled to the codec;and a digital interface coupled to the channel adaptor and furthercoupled to the remote device via the connection, wherein the channeladaptor is operable to convert the encoded audio input for transmissionthrough the digital interface and the connection.
 19. The system ofclaim 18, wherein the connection is an IP connection, an ISDN connectionor a DSL connection.
 20. The system of claim 18, wherein the connectionmanagement module in the local communication device includes: an audioquality feedback module coupled to the local interface; a data ratemodule coupled to the local interface; a data transmission standardmodule coupled to the local interface; an audio bandwidth adjustmentmodule coupled to the local interface; and a modem training modulecoupled to the local interface and the communication interface module.